Commit 8bd93c99 authored by Sebastian Dröge's avatar Sebastian Dröge 🍵

Release 1.2.1

parent 079f2498
=== release 1.2.1 ===
2013-11-09 Sebastian Dröge <slomo@coaxion.net>
* configure.ac:
releasing 1.2.1
2013-11-09 12:01:11 +0100 Sebastian Dröge <sebastian@centricular.com>
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/el.po:
* po/en_GB.po:
* po/eo.po:
* po/es.po:
* po/eu.po:
* po/fi.po:
* po/fr.po:
* po/gl.po:
* po/hr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/lv.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ro.po:
* po/ru.po:
* po/sk.po:
* po/sl.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
po: Update translations
2013-10-16 16:46:05 -0300 Thibault Saunier <thibault.saunier@collabora.com>
* gst/playback/gstrawcaps.h:
playback: Add subpicture/x-dvb as raw caps
https://bugzilla.gnome.org/show_bug.cgi?id=710325
2013-11-07 15:03:34 +0000 Tom Greenwood <tcdgreenwood@hotmail.com>
* gst-libs/gst/app/gstappsrc.c:
appsrc: Fix deadlock that may occur when multiple threads access appsrc at once
https://bugzilla.gnome.org/show_bug.cgi?id=711550
2013-11-01 17:02:22 +0100 Sebastian Dröge <sebastian@centricular.com>
* docs/libs/gst-plugins-base-libs-sections.txt:
* win32/common/libgstrtsp.def:
rtspconnection: Add new API to the docs and .def file
2013-11-01 16:43:56 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst-libs/gst/rtsp/gstrtspconnection.h:
rtspconnection: Fix indention in header
2013-11-01 07:25:01 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspconnection.h:
rtspconnection: allow setting tls certificate validation
Added new functions gst_rtsp_connection_set_tls_validation_flags() to
allow setting the TLS certificate validation flags when establishing a
TLS connection.
A getter is also available, gst_rtsp_connection_get_tls_validation_flags().
https://bugzilla.gnome.org/show_bug.cgi?id=711231
2013-10-28 12:36:04 +0100 Antonio Ospite <ospite@studenti.unina.it>
* gst/videoscale/gstvideoscale.c:
videoscale: fix adding borders when NV12 is used
When the frame buffer is NV12 the borders are not added at all, fix that
and fill them to black.
https://bugzilla.gnome.org/show_bug.cgi?id=711003
2013-10-07 22:51:04 +0200 Stefan Sauer <ensonic@users.sf.net>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: early return when we have no streams
2013-10-07 22:51:46 +0200 Stefan Sauer <ensonic@users.sf.net>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: don't shadow local variables
2013-10-14 18:45:16 +0200 Stefan Sauer <ensonic@users.sf.net>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: also filter 'framed' field when looking for same streams
Fixes extra streams for some mp4 files containing aac audio.
2013-10-07 22:52:27 +0200 Stefan Sauer <ensonic@users.sf.net>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: filter 'parsed' field when checking for same caps
We're checking the caps to see if we got more caps details after a parser got
plugged. This will also have a flipped 'parsed' field. If the field was already
present before the parse the match will fail. Add a function that will do the
check while excluding this field.
2013-10-11 21:51:00 +0200 Stephan Sundermann <stephansundermann@gmail.com>
* gst-libs/gst/video/navigation.c:
navigation: Add missing out parameter annotations to GstNavigation
https://bugzilla.gnome.org/show_bug.cgi?id=709938
2013-10-08 16:02:46 +0200 Takashi Iwai <tiwai@suse.de>
* gst-libs/gst/audio/gstaudioringbuffer.c:
audioringbuffer: Don't clear need_reorder flag too early
gst_audio_ring_buffer_set_channel_positions() checks whether the given
positions are identical with the current setup and returns
immediately if so. But it also clears need_reorder flag before this
comparison, thus this flag might be wrongly cleared if the function is
called twice with the same channel positions.
Move the flag clearance after the check.
https://bugzilla.gnome.org/show_bug.cgi?id=709754
2013-10-08 09:13:50 +0200 Stefan Sauer <ensonic@users.sf.net>
* gst-libs/gst/video/gstvideodecoder.c:
videodecoder: don't overflow in bytes<->time conversion
fps_n and _d values can be large and this can overflow a uint. Also fix
copy'n'paste mistake in comments.
2013-10-04 13:57:51 +0200 Matej Knopp <matej.knopp@gmail.com>
* gst/audioconvert/gstaudioconvert.c:
audioconvert: Map buffer as READWRITE if the buffer and memory is writable
and only use the input buffer as temporary buffer in that case.
https://bugzilla.gnome.org/show_bug.cgi?id=709408
2013-10-02 15:02:44 +0200 Sebastian Dröge <slomo@circular-chaos.org>
* gst/playback/gstplaysink.c:
playsink: If the visualisation is changing and reconfiguration is pending, do it all during reconfiguration
Otherwise we will have two pad blocks that want to use the same mutex
and block each other via the streamlock.
https://bugzilla.gnome.org/show_bug.cgi?id=709210
2013-10-08 16:13:58 -0300 Thiago Santos <ts.santos@partner.samsung.com>
* tests/check/elements/videotestsrc.c:
videotestsrc: improve test for backwards playback
Improve test by checking that timestamps are decreasing
2013-06-07 16:32:23 -0400 Thibault Saunier <thibault.saunier@collabora.com>
* tests/check/elements/videotestsrc.c:
tests: test videotestsrc in reverse playback
https://bugzilla.gnome.org/show_bug.cgi?id=701813
2013-10-08 00:08:34 -0300 Thiago Santos <ts.santos@partner.samsung.com>
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
videotestsrc: implement reverse playback
Decrement the n_frames counter when doing reverse playback to
have timestamps and offsets reducing instead of increasing
https://bugzilla.gnome.org/show_bug.cgi?id=701813
2013-09-24 16:47:52 -0700 Thiago Santos <ts.santos@partner.samsung.com>
* gst/playback/gstplaybin2.c:
playbin: make sure elements are in null before disposing
If a pipeline fails to preroll, it might happen that the sinks are
put into READY state from playbin's sink activation, but they are never
set to playsink, so they aren't being managed by a GstBin and will keep
their READY state until they are unreffed, leading to a warning.
Prevent this by always forcing them to NULL when deactivating a group
https://bugzilla.gnome.org/show_bug.cgi?id=708789
2013-09-30 21:46:10 +0200 Hans Månsson <hansm@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtspconnection: Connect to proxy if specified
Reference: https://bugzilla.gnome.org/show_bug.cgi?id=708880
2013-09-27 22:41:28 +0200 Matej Knopp <matej.knopp@gmail.com>
* gst/audiorate/gstaudiorate.c:
audiorate: clip buffer before pushing it
https://bugzilla.gnome.org/show_bug.cgi?id=708953
2013-09-27 22:40:28 +0200 Matej Knopp <matej.knopp@gmail.com>
* gst-libs/gst/audio/audio.c:
audio: change buffer timestamp when clipping even if data hasn't been trimmed
https://bugzilla.gnome.org/show_bug.cgi?id=708952
2013-09-27 22:53:43 +0200 Matej Knopp <matej.knopp@gmail.com>
* gst-libs/gst/pbutils/descriptions.c:
pbutils: Add entry for text/x-raw
https://bugzilla.gnome.org/show_bug.cgi?id=708954
2013-09-25 19:29:24 +0200 Matej Knopp <matej.knopp@gmail.com>
* gst-libs/gst/pbutils/descriptions.c:
pbutils: add MPEG 2 AAC description
https://bugzilla.gnome.org/show_bug.cgi?id=708773
2013-09-24 16:26:37 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtspconnection: Unset input/output_stream after freeing the GIOStream
watch->input_stream and watch->output_stream are owned by the GIOStream
and should be unset after freeing the stream.
https://bugzilla.gnome.org/show_bug.cgi?id=708689
2013-09-24 17:24:05 +0100 Tim-Philipp Müller <tim@centricular.net>
* README:
* common:
Automatic update of common submodule
From 6b03ba7 to 7412249
=== release 1.2.0 ===
2013-09-24 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2013-09-24 14:16:22 +0200 Sebastian Dröge <slomo@circular-chaos.org>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
releasing 1.2.0
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-encoding.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-ivorbisdec.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-videoconvert.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst-plugins-base.doap:
* win32/common/_stdint.h:
* win32/common/config.h:
Release 1.2.0
2013-09-24 14:14:18 +0200 Sebastian Dröge <slomo@circular-chaos.org>
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/el.po:
* po/en_GB.po:
* po/eo.po:
* po/es.po:
* po/eu.po:
* po/fi.po:
* po/fr.po:
* po/gl.po:
* po/hr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/lv.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ro.po:
* po/ru.po:
* po/sk.po:
* po/sl.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
Update .po files
2013-09-24 12:47:26 +0200 Sebastian Dröge <slomo@circular-chaos.org>
This is GStreamer Base Plugins 1.2.0
Changes since 1.0:
New API:
• GstContext negotiation / sharing / announcing for sharing a
generic context between elements, e.g. a display handle
• GL texture upload conversion meta for allowing different
buffer types to be converted to an OpenGL texture
• GstCapsFeatures as extension to GstCaps for allowing the
negotiation of specific memory or meta requirements between
elements
• GstMemory flags for contiguous and non-mappable memory
• The stream-start event has optional flags now, e.g. for signalling
sparse streams
• The stream-start even has an optional group-id field now to signal
all streams that should be played together
• Allocators library in gst-plugins-base, currently only with generic
dmabuf memory support
• insertbin library for easier handling of dynamically linked
pipelines (in -bad for now)
• EGL helper library (in -bad for now)
• MPEG-TS data structure library (in -bad for now)
• New GstVideoRegionOfInterestMeta to describe a region of interest on
video frames.
• GstVideoDecoder/Encoder has new ::flush() vfunc to replace the
ill-defined ::reset() vfunc.
• The URI query allows to query the redirected URI now.
Major changes:
• New tool: gst-play-1.0 in gst-plugins-base for basic playback
testing on the command line.
• New plugins:
∘ mssdemux for Microsoft Smooth Streaming
∘ dashdemux for DASH adaptive streaming protocol
∘ bluez for interaction with Bluetooth devices
∘ openjpeg for JPEG2000 decoding and encoding
∘ daala for experimental Daala decoding and encoding
∘ vpx plugin has experimental VP9 decoding and encoding support
∘ webp plugin for WebP decoding (encoding to be added later)
∘ Various others: yadif, srtp, sbc, fluidsynth, midiparse,
mfc, ivtv, accuraterip and audiofxbad
• Moved plugins:
∘ dtmf, vp8rtp, scaletempo and rtpmux plugins are in
gst-plugins-good now
• Video:
∘ Fix handling of interlaced video in converters such as videoscale
and videoconvert (e.g. scale both fields independently)
∘ videoconvert will try harder to minimise quality losses when
conversion is necessary
∘ The experimental GstSurfaceConverter, GstSurfaceMeta and
GstVideoContext APIs from the (confusingly-named)
libgstbasevideo-1.0 library in gst-plugins-bad have now been
removed and been replaced by new APIs in GStreamer Core and
gst-plugins-base (see above). Since that was all that was left in
this library, the entire experimental libgstbasevideo-1.0 library
has been removed from gst-plugins-bad
∘ Chroma subsampling and chroma siting conversion is better handled
in videoconvert and the support for interlaced video was improved.
∘ New pinwheel and spoke patterns in videotestsrc
∘ videomixer can now accept different video formats on its sinkpads
and converts to a common format during mixing
• Audio:
∘ audioconvert will try harder to minimise quality losses when
conversion is necessary
∘ adder now allows muting/unmuting of its input streams, and also
per-input stream volume
∘ pulseaudio elements can switch between devices during playback now
∘ aacparse can convert between ADTS←→RAW
• Platform specific changes:
∘ Caps, events, etc. are now printed in the GStreamer debug logs
with their content instead of just the pointer address even on
non-glibc platforms (e.g. Windows, OSX, Android).
∘ Network elements (UDP/TCP) now work better with platforms,
where IPv6 sockets can't handle IPv4 (e.g. Windows)
∘ Linux/BSD: v4l2 had many improvements and cleanups
• Other changes:
∘ gst-libav now uses libav 9
∘ Static linking of plugins is supported now (also in 1.0.7)
∘ rtspsrc: add support for NetClientClock: when the server suggests a
GstNetTimeProvider in the SDP, set up a GstNetClientClock that
slaves to the remote clock and suggest this clock in provide_clock.
Simplifies synchronized playback of a resource from an RTSP server.
gst-rtsp-server now supports adding this to the SDP and can provide
a network clock
∘ RTP retransmission / NACK support and big RTP jitterbuffer improvements
∘ SRTP and DTLS support
∘ Changes to many elements and core to use the correct sticky event
order and also not lose any important sticky events during flushing
∘ >1000 fixed bug reports, and many other bug fixes and other
improvements everywhere that had no bug report
Things to look out for:
• Single header includes for all libraries, e.g. #include
<gst/video/video.h> - this was needed for some bindings.
• Stricter (correct) caps subset checking in some cases where this was
not correct before. Caps will now always fail to be a compatible
subset of another set of caps if the subset caps are missing some
fields that the superset caps have. This might lead to not-negotiated
errors if caps are incomplete now. However, it also prevents possible
data corruption caused by piping data formatted in an
incompatible/unexpected way into some elements. Check your h264 caps
for stream-format and alignment fields and AAC caps for the
stream-format field. This change will also be included in the next
stable 1.0.8 release.
• Stricter checking for missing events and correct sticky event order
(stream-start, caps, segment) in some places; this is not enabled in
stable releases by default, but you may get warnings when using git
builds, development releases or when compiling with
-UG_DISABLE_ASSERT in CFLAGS
• x264enc now outputs data in byte-stream by default if downstream has
ANY caps (e.g. appsink without caps set, filesink, udpsink,
tcpserversink etc.)
• The MPEG TS demuxer posts messages contain the PMT, PAT, etc. in a
different format now. This new format uses the data structures from
the new MPEGTS library
• The GstContext API has changed between 1.1.4 and 1.1.90
This is GStreamer Base Plugins 1.2.1
Release notes for GStreamer Base Plugins 1.2.0
Release notes for GStreamer Base Plugins 1.2.1
The GStreamer team is proud to announce a new feature release
The GStreamer team is proud to announce a new bug-fix release
in the 1.x stable series of the
core of the GStreamer streaming media framework.
......@@ -60,10 +60,23 @@ contains a set of codecs plugins based on libav (formerly gst-ffmpeg)
Bugs fixed in this release
* 708667 : rtspconnection: leaks file descriptors/child sources
* 708372 : dmabuf: sys/mman.h: No such file or directory
* 708590 : adder: Should send its segment before checking for eos
* 708606 : video-frame: offsets are not copied from metadata
* 701813 : Reverse playback not working with videotestsrc
* 708689 : rtspconnection: RTSP watch is dispatched after closing the connection
* 708773 : pbutils: add MPEG 2 AAC description
* 708789 : playbin: make sure elements are in null before disposing
* 708880 : rtspconnection: Not connecting to proxy when specified
* 708952 : audio: change buffer ts when clipping buffer even if data length is same
* 708953 : audiorate: clip buffers before pushing them out
* 708954 : pbutils: add entry for text/x-raw
* 709210 : Hangs on startup getting PulseAudio volume
* 709408 : audioconvert: modifies buffer mapped for READ
* 709637 : oggmux: Make sure we end up sending EOS if we received EOS on all sinkpads
* 709754 : audioringbuffer: Clears need_reorder flag wrongly
* 709938 : navigation: Missing gobject-introspection annotations
* 710325 : playback: Add subpicture/x-dvb as raw caps
* 711003 : videoscale: borders are filled with green when using NV12 pixelformat
* 711231 : rtspconnection: allow setting tls certificate validation flags
* 711550 : appsrc: Deadlocking because holding mutex while setting caps
==== Download ====
......@@ -100,10 +113,17 @@ subscribe to the gstreamer-devel list.
Contributors to this release
* Edward Hervey
* Mathieu Duponchelle
* Aleix Conchillo Flaque
* Antonio Ospite
* Hans Månsson
* Matej Knopp
* Ognyan Tonchev
* Sebastian Dröge
* Stefan Sauer
* Stephan Sundermann
* Takashi Iwai
* Thiago Santos
* Thibault Saunier
* Tim-Philipp Müller
* Wim Taymans
* Tom Greenwood
 
\ No newline at end of file
......@@ -5,7 +5,7 @@ dnl please read gstreamer/docs/random/autotools before changing this file
dnl initialize autoconf
dnl releases only do -Wall, git and prerelease does -Werror too
dnl use a three digit version number for releases, and four for git/prerelease
AC_INIT([GStreamer Base Plug-ins],[1.2.0],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-base])
AC_INIT([GStreamer Base Plug-ins],[1.2.1],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-base])
AG_GST_INIT
......@@ -56,7 +56,7 @@ dnl 1.2.5 => 205
dnl 1.10.9 (who knows) => 1009
dnl
dnl sets GST_LT_LDFLAGS
AS_LIBTOOL(GST, 200, 0, 200)
AS_LIBTOOL(GST, 201, 0, 201)
dnl *** required versions of GStreamer stuff ***
GST_REQ=1.2.0
......
......@@ -3,7 +3,7 @@
<description>Adds multiple streams</description>
<filename>../../gst/adder/.libs/libgstadder.so</filename>
<basename>libgstadder.so</basename>
<version>1.2.0</version>
<version>1.2.1</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......
......@@ -3,7 +3,7 @@
<description>ALSA plugin library</description>
<filename>../../ext/alsa/.libs/libgstalsa.so</filename>
<basename>libgstalsa.so</basename>
<version>1.2.0</version>
<version>1.2.1</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......
......@@ -3,7 +3,7 @@
<description>Elements used to communicate with applications</description>
<filename>../../gst/app/.libs/libgstapp.so</filename>
<basename>libgstapp.so</basename>
<version>1.2.0</version>
<version>1.2.1</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......
......@@ -3,7 +3,7 @@
<description>Convert audio to different formats</description>
<filename>../../gst/audioconvert/.libs/libgstaudioconvert.so</filename>
<basename>libgstaudioconvert.so</basename>
<version>1.2.0</version>
<version>1.2.1</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......
......@@ -3,7 +3,7 @@
<description>Adjusts audio frames</description>
<filename>../../gst/audiorate/.libs/libgstaudiorate.so</filename>
<basename>libgstaudiorate.so</basename>
<version>1.2.0</version>
<version>1.2.1</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......
......@@ -3,7 +3,7 @@
<description>Resamples audio</description>
<filename>../../gst/audioresample/.libs/libgstaudioresample.so</filename>
<basename>libgstaudioresample.so</basename>
<version>1.2.0</version>
<version>1.2.1</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......
......@@ -3,7 +3,7 @@
<description>Creates audio test signals of given frequency and volume</description>
<filename>../../gst/audiotestsrc/.libs/libgstaudiotestsrc.so</filename>
<basename>libgstaudiotestsrc.so</basename>
<version>1.2.0</version>
<version>1.2.1</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......
......@@ -3,7 +3,7 @@
<description>Read audio from CD in paranoid mode</description>
<filename>../../ext/cdparanoia/.libs/libgstcdparanoia.so</filename>
<basename>libgstcdparanoia.so</basename>
<version>1.2.0</version>
<version>1.2.1</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......
......@@ -3,7 +3,7 @@
<description>various encoding-related elements</description>
<filename>../../gst/encoding/.libs/libgstencodebin.so</filename>
<basename>libgstencodebin.so</basename>
<version>1.2.0</version>
<version>1.2.1</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......
......@@ -3,7 +3,7 @@
<description>GIO elements</description>
<filename>../../gst/gio/.libs/libgstgio.so</filename>
<basename>libgstgio.so</basename>
<version>1.2.0</version>
<version>1.2.1</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......
......@@ -3,7 +3,7 @@
<description>Vorbis Tremor decoder</description>
<filename>../../ext/vorbis/.libs/libgstivorbisdec.so</filename>
<basename>libgstivorbisdec.so</basename>
<version>1.2.0</version>
<version>1.2.1</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......
......@@ -3,7 +3,7 @@
<description>libvisual visualization plugins</description>
<filename>../../ext/libvisual/.libs/libgstlibvisual.so</filename>
<basename>libgstlibvisual.so</basename>
<version>1.2.0</version>
<version>1.2.1</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......
......@@ -3,7 +3,7 @@
<description>ogg stream manipulation (info about ogg: http://xiph.org)</description>
<filename>../../ext/ogg/.libs/libgstogg.so</filename>
<basename>libgstogg.so</basename>
<version>1.2.0</version>
<version>1.2.1</version>